Thanks Sherwood for all the info.<div><br></div><div>The devices are using ulaw and rfc2833. There is no transcoding on my server, but not sure what my trunk providers are doing.<br><br></div><div>I was thinking about the frequency detection issue as it seems to be primarily involving women so I'll try adjusting the input/output gain to see if that helps. I've tried all other combinations of DTMF settings on the ATAs so that's my last hope on the device end.</div>
<div><br></div><div>I'm hoping to avoid the packet capture as that is never a fun road to go down but that is probably the next step.</div><div><br></div><div>Any idea why Asterisk still ques/replays the tone considering it see's it's shorter than the 80s minimum?</div>
<div><br></div><div>Travis</div><div><br></div><div>Hoping to avoid<br><div class="gmail_quote">On Wed, Jul 28, 2010 at 7:16 PM, Sherwood McGowan <span dir="ltr"><<a href="mailto:sherwood.mcgowan@gmail.com">sherwood.mcgowan@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">Sorry, I came into this late...what codec is the device using, and is<br>
the audio being trascoded?<br>
<br>
Back at Voxitas, we had a couple of customers complain about random<br>
DTMF tones coming across their line, and Asterisk WAS actually<br>
"hearing" DTMF tones...want to know what it was?.....<br>
<br>
In that particular case (just a place to start looking) it was G729 on<br>
customer ATAs (don't remember the models)....Here's the freaky<br>
thing....It only happened with CERTAIN people talking on the<br>
phone...IIRC, we determined that the ATA's G729 processor was<br>
mistaking certain audio frequencies in the speaker's voice and<br>
believing it was a DTMF tone from the analog device and sending the<br>
appropriate DTMF signal to our servers...<br>
<br>
I'm sorry, I don't remember how we fixed it...I think we did some<br>
audio tweaking (advanced ATA config, input level, out level, etc..),<br>
be we may have just ended up having to tell that client to not use<br>
G729 on those ATAs....<br>
<br>
This _MAY_ happen with other codecs, but I think it's mainly either<br>
G729..maybe primarily transcoding?<br>
<br>
<br>
NERDY FUn Crap below:<br>
<br>
capture SIP and RTP between your Asterisk and an offending device<br>
(writing to a file)....then start doing everything you can to cause<br>
the DTMF issue to occur. NOW, open your capture in wireshark...dump<br>
the RTP payload to a file and open that file in an audio editor....<br>
<br>
Now, go through the wireshark capture...see if you see any DTMF events<br>
(if rfc2833 it'll be an RTP EVENT, if SIP INFO, it'll be a sip info,<br>
and if you're using inband **SHUDDER** you can just listen to the<br>
audio).....note the time in seconds from the beginning of the audio<br>
stream whenever a DTMF event occurs, and then go to that spot in the<br>
audio file....If you're feeling REALLY frisky, do a frequency<br>
analysis...I'll bet you'll see that the voice that is speaking at the<br>
time of the DTMF event on your various captures will have a frequency<br>
range in common...a very close range...maybe look up DTMF tone<br>
definition and get the freqs....(did it....more detail than even I<br>
feel like doing right now :D)<br>
<br>
Cheers,<br>
Sherwood McGowan<br>
<div><div></div><div class="h5"><br>
On Wed, Jul 28, 2010 at 6:43 PM, Travis Langhals <<a href="mailto:travis@netitek.com">travis@netitek.com</a>> wrote:<br>
> SIP/5211 is a Grandstream device.<br>
> Did not add relaxdtmf=no, but sip show settings verifies it's already set to<br>
> no.<br>
> Fat fingered the version, it should have said 1.6.2.6 through 1.6.2.10<br>
> Travis<br>
><br>
> On Wed, Jul 28, 2010 at 3:12 AM, Benny Amorsen <<a href="mailto:benny%2Busenet@amorsen.dk">benny+usenet@amorsen.dk</a>><br>
> wrote:<br>
>><br>
>> Travis Langhals <<a href="mailto:travis@netitek.com">travis@netitek.com</a>> writes:<br>
>><br>
>> > [2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF begin '1' received on<br>
>> > SIP/5211-00000078<br>
>><br>
>> Is SIP/5211 a Linksys or a Grandstream or something else?<br>
>><br>
>> Do you have relaxdtmf=no?<br>
>><br>
>> Also, your Asterisk version numbers are incorrect. Do you mean 1.6.2.10?<br>
>><br>
>><br>
>> /Benny<br>
>><br>
><br>
><br>
</div></div>> --<br>
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